Really strange answer. I am non used to search on playboy.com. Anyway, if you try to search insecure=very on www.voip-info.org, you find 742 links , a bit more for me. (I just want to know what it means)
Moreovere, the first 20 links are non accessible at all http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+sip+insecure&diff=6 they speak about tiki-pagehistory.php, which appears not to exist. no other comments about this. ************************************************************ I know about one project , "asterisk documentation project" http://www.asteriskdocs.org in its home page, the first line is Great software needs great documentation. I really hope this project will be implemented, without documentation evrything is too hard Andrea "Steve Totaro" <[EMAIL PROTECTED] echnologies.com> To Sent by: "Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion" [EMAIL PROTECTED] <asterisk-users@lists.digium.com> m.com cc Subject 12/10/2005 14.53 Re: [Asterisk-Users] parameters documentation Please respond to Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED] ists.digium.com> www.voip-info.org > Another trivial question: > > Is there a "place" where all the parameters are documented ? > In example (my example!) I would like to know the meaning of a lot of > parameter that can be used in sip.conf, > > A lot of these keywords are intuitive keywords (i.e. NAT=YES/NO;PORT=5060; > context=xxxxx) but other are not (at least for me) > i.e.: > > type = peer, friend > insecure=very > host=dynamic > > and so on. > > At last, my need is: > > Accept a non-registerd sip-strem from a well known ip address (and only > from that ip address....) > > I tried to add a > > [testsip] > ;username=testsip > type=friend > ;secret=testsip > qualify=no > port=5060 > nat=no > host=x.y.z.w > dtmfmode=rfc2833 > context=from-internal > canreinvite=no > callerid="test sip " <testsip> > > that would work if the sip would be registered. But the SIP client is not > able to register. > > I solved using the > context = from-sip-external ; Send unknown SIP callers to this context > > and it works, but I have no more the control about who is sending me SIP > stream (anybody now can use my asterisk box...) > > any help will be greatly appreciated > > Andrea _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users